[Problem] Problem den Provider GulfSip zu konfigurieren

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MET

MET

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Wollte den für mich neuen Provider GuflSip ausprobieren. Mit einem Softphone funktionierte dies auf anhieb. Aus irgend einem Grund krieg ich diesen Provider auf dem Asterisk nicht hin. Nachfolgend den Eintrag in Sip.conf und die Meldungen unter sip debug von einem Verbindungsaufbau. Hat jemand eine Ahnung wo das Problem liegen könnte und vielleicht einen Vorschlag wie dies behoben werden könnte? Liegt es vielleicht am speziellen Port den dieser Provider verwendet?

sip.conf:
PHP: 
[general]
alwaysauthreject=yes
context=default
bindport=5060
bindaddr=IP_von_Asterisk
srvlookup=yes
useragent=MyDevice
tos_audio=0xb0
tos_sip=0xb0
disallow=all
allow=ulaw
allow=alaw
allow=ilbc

register => User_Nr:[email protected]:6321/GulfSip

[GulfSip]
type=peer
port=6321
username=User_Nr
fromuser=+...
secret=geheim
host=sip.gulfsip.com
nat=yes
insecure=port,invite
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=ilbc

[GulfSip_in]
type=peer
host=sip.gulfsip.com
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
context=incoming

Debug von Verbindungsaufbau:
PHP: 
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
vs8709*CLI>
<--- SIP read from IP_von_Lynksys:5061 --->
NOTIFY sip:IP_von_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030
From: <sip:10b@IP_von_Asterisk>;tag=210a5427b3032177o1
To: <sip:IP_von_Asterisk>
Call-ID: [email protected]
CSeq: 585 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.42 : 5061 (no NAT)
vs8709*CLI>
<--- Transmitting (no NAT) to 192.168.1.42:5061 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030;received=IP_von_Lynksys
From: <sip:10b@IP_von_Asterisk>;tag=210a5427b3032177o1
To: <sip:IP_von_Asterisk>;tag=as73692944
Call-ID: [email protected]
CSeq: 585 NOTIFY
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
vs8709*CLI>
<--- SIP read from IP_von_Lynksys:5061 --->
NOTIFY sip:IP_von_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030
From: <sip:10b@IP_von_Asterisk>;tag=210a5427b3032177o1
To: <sip:IP_von_Asterisk>
Call-ID: [email protected]
CSeq: 585 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.42 : 5061 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.42:5061 --->
IP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.42:5061;branch=z9hG4bK-ce959030;received=IP_von_Lynksys
From: <sip:10b@IP_von_Asterisk>;tag=210a5427b3032177o1
To: <sip:IP_von_Asterisk>;tag=as5a33e10b
Call-ID: [email protected]
CSeq: 585 NOTIFY
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
vs8709*CLI>
<--- SIP read from IP_von_Lynksys:5062 --->
NOTIFY sip:IP_von_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-6baf9108
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=829651a7b1eb41f7o0
To: <sip:IP_von_Asterisk>
Call-ID: [email protected]
CSeq: 575 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.42 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.42:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-6baf9108;received=IP_von_Lynksys
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=829651a7b1eb41f7o0
To: <sip:IP_von_Asterisk>;tag=as2d786770
Call-ID: [email protected]
CSeq: 575 NOTIFY
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
vs8709*CLI>
<--- SIP read from IP_von_Lynksys:5062 --->
INVITE sip:00NR_anzurufen@IP_von_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-1c79e8f0
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=d90b1a2ca4f54990o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>
Remote-Party-ID: "+NR_anzurufen" 

<sip:10a@IP_von_Asterisk>;screen=yes;party=calling
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "+NR_anzurufen" <sip:[email protected]:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 1198925 1198925 IN IP4 192.168.1.42
s=-
c=IN IP4 192.168.1.42
t=0 0
m=audio 16460 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (15 headers 20 lines) ---
Sending to 192.168.1.42 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]

<--- Reliably Transmitting (NAT) to IP_von_Lynksys:5062 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-1c79e8f0;received=IP_von_Lynksys
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=d90b1a2ca4f54990o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>;tag=as149d3d12
Call-ID: [email protected]
CSeq: 101 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="13829fd9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms 

(Method: INVITE)
Found user '10a'
vs8709*CLI>
<--- SIP read from IP_von_Lynksys:5062 --->
ACK sip:00NR_anzurufen@IP_von_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-1c79e8f0
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=d90b1a2ca4f54990o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>;tag=as149d3d12
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "+NR_anzurufen" <sip:[email protected]:5060>
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
vs8709*CLI>
<--- SIP read from IP_von_Lynksys:5062 --->
INVITE sip:00NR_anzurufen@IP_von_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e0ee288c
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=d90b1a2ca4f54990o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>
Remote-Party-ID: "+NR_anzurufen" 

<sip:10a@IP_von_Asterisk>;screen=yes;party=calling
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest 

username="10a",realm="asterisk",nonce="13829fd9",uri="sip:00NR_anzurufen@IP_von_As

terisk",algorithm=MD5,response="61794f75ff9ee497e0d1967177766998"
Contact: "+NR_anzurufen" <sip:[email protected]:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 1198925 1198925 IN IP4 192.168.1.42
s=-
c=IN IP4 192.168.1.42
t=0 0
m=audio 16460 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (16 headers 20 lines) ---
Sending to IP_von_Lynksys : 5062 (NAT)
Using INVITE request as basis request - [email protected]
Found user '10a'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x90d (g723|

ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-

event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.42:16460
Looking for 00NR_anzurufen in app10 (domain IP_von_Asterisk)
list_route: hop: <sip:[email protected]:5060>
vs8709*CLI>
<--- Transmitting (NAT) to IP_von_Lynksys:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e0ee288c;received=IP_von_Lynksys
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=d90b1a2ca4f54990o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:00NR_anzurufen@IP_von_Asterisk>
Content-Length: 0


<------------>
    -- Executing [00NR_anzurufen@app10:1] Dial("SIP/10a-0000074f", 

"SIP/00NR_anzurufen@GulfSip|45|r") in new stack
Audio is at IP_von_Asterisk port 10276
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 37.58.67.5:6321:
INVITE sip:[email protected]:6321 SIP/2.0
Via: SIP/2.0/UDP IP_von_Asterisk:5060;branch=z9hG4bK6d173508;rport
From: "Home" <sip:+NR_fromuser@IP_von_Asterisk>;tag=as52572b92
To: <sip:[email protected]:6321>
Contact: <sip:+NR_fromuser@IP_von_Asterisk>
Call-ID: 1044041b2cb5783f01d1d6371d0ab938@IP_von_Asterisk
CSeq: 102 INVITE
User-Agent: MyDevice
Max-Forwards: 70
Date: Fri, 14 Sep 2012 15:49:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 16787 16787 IN IP4 IP_von_Asterisk
s=session
c=IN IP4 IP_von_Asterisk
t=0 0
m=audio 10276 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendr

===================================================
* * * * * * Fortsetzung von anderm Test * * * * * *
===================================================

v=0
o=root 16787 16787 IN IP4 IP_von_Asterisk
s=session
c=IN IP4 IP_von_Asterisk
t=0 0
m=audio 18412 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 00NR_anzurufen@GulfSip
vs8709*CLI>
<--- Transmitting (NAT) to IP_von_Lynksys:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e6291fe8;received=IP_von_Lynksys
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=e335ad4a72f4390o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>;tag=as099fc966
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:00NR_anzurufen@IP_von_Asterisk>
Content-Length: 0


<------------>
vs8709*CLI>
<--- SIP read from 37.58.67.5:6321 --->
SIP/2.0 403 No relaying
Via: SIP/2.0/UDP 

IP_von_Asterisk:5060;received=IP_von_Asterisk;branch=z9hG4bK31b7ebf3;rport=5060
From: "Home" <sip:+NR_fromuser@IP_von_Asterisk>;tag=as5591a135
To: 

<sip:[email protected]:6321>;tag=3c0de42e4be99a5add523ae5e2615b5d.503

0
Call-ID: 299870370ba8a8e93cddf33973e3d32c@IP_von_Asterisk
CSeq: 102 INVITE
Server: OpenSIPS (1.7.1-tls (x86_64/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 37.58.67.5:6321:
ACK sip:[email protected]:6321 SIP/2.0
Via: SIP/2.0/UDP IP_von_Asterisk:5060;branch=z9hG4bK31b7ebf3;rport
From: "Home" <sip:+NR_fromuser@IP_von_Asterisk>;tag=as5591a135
To: 

<sip:[email protected]:6321>;tag=3c0de42e4be99a5add523ae5e2615b5d.503

0
Contact: <sip:+NR_fromuser@IP_von_Asterisk>
Call-ID: 299870370ba8a8e93cddf33973e3d32c@IP_von_Asterisk
CSeq: 102 ACK
User-Agent: MyDevice
Max-Forwards: 70
Content-Length: 0


---
[2012-09-14 17:45:06] WARNING[17168]: chan_sip.c:13053 handle_response_invite: 

Received response: "Forbidden" from '"Home" <sip:

+NR_fromuser@IP_von_Asterisk>;tag=as5591a135'
    -- SIP/GulfSip-0000074e is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/10a-0000074d' status is 'CONGESTION'
vs8709*CLI>
<--- Reliably Transmitting (NAT) to IP_von_Lynksys:5062 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e6291fe8;received=IP_von_Lynksys
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=e335ad4a72f4390o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>;tag=as099fc966
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21


<------------>
Really destroying SIP dialog '299870370ba8a8e93cddf33973e3d32c@IP_von_Asterisk' 

Method: INVITE
vs8709*CLI>
<--- SIP read from IP_von_Lynksys:5062 --->
ACK sip:00NR_anzurufen@IP_von_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-e6291fe8
From: "+NR_anzurufen" <sip:10a@IP_von_Asterisk>;tag=e335ad4a72f4390o0
To: <sip:00NR_anzurufen@IP_von_Asterisk>;tag=as099fc966
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest 

username="10a",realm="asterisk",nonce="17bd5247",uri="sip:00NR_anzurufen@IP_von_As

terisk",algorithm=MD5,response="5a874f03246b9bc85a1fd2753f23902f"
Contact: "+NR_anzurufen" <sip:[email protected]:5060>
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
vs8709*CLI>
 
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